Voice over Internet Protocol (VoIP) services allow IP telephones to communicate with each other over the Internet. However, an IP telephone may also be used to communicate through with traditional telephones through the PSTN. In order to do this, a gateway is typically used to convert the voice data packets from the IP telephone to a TDM stream using a 64 kb/s PCM format so that it may traverse the PSTN and be received, and understood, by a traditional telephone.
Two IP telephones may also communicate over the PSTN if one calls the other one using a standard telephone number, such as an NPA-NXX number, and where such communication is enabled by a pair of IP/PSTN gateways. This type of PSTN mediated call may be required in cases where the IP capability or addressability of the other VoIP endpoint is not known, which is often the case where the administrative domains of the VoIP endpoints differ. If a VoIP endpoint does not know what type of phone is at the other end of the communication, it must assume that the other endpoint is a traditional telephone which requires normal 64 kb/s TDM communication through the PSTN, and so conversion from packet voice data to 64 kb/s PCM is employed. IP telephones will often use one of several standardized voice compression algorithms before packetization of the voice stream. These algorithms are known as “lossy” since each successive compression and decompression will result in increased latency and reduced voice quality. A call involving two IP telephones over the PSTN can then involve an initial compression of the voice stream, a decompression (to 64 kb/s PCM), and then a recompression by the distant gateway to the IP endpoint. This reduces the quality of the communication to below that which could be otherwise be achieved if the two IP telephones recognize that each other is capable of packet based communications, and could communicate directly in the packet domain without requiring translation to confines of 64 kb/s PCM voice.
A mechanism which allowed IP telephones to communicate directly with each other over the PSTN in the packet domain while still allowing interworking of IP telephones with the legacy PSTN would improve speech quality, and also enable support of functionality beyond simple voice.